High Load Diminution by Regulating Timers in SIP Servers

To start voice, image, instant messaging, and generally multimedia communication, session communication must begin between two participants. SIP (session initiation protocol) that is an application layer control induces management and terminates this kind of sessions. As far as the independence of SIP from transport layer protocols is concerned, SIP messages can be transferred on a variety of transport layer protocols including TCP or UDP. Mechanism of Retransmission that is embedded in SIP could compensate for the missing packet loss, in case of need. This mechanism is applied when SIP messages are transmitted on an unreliable transmission layer protocol like UDP. Also, while facing SIP proxy with overload, it could cause excessive filling of proxy queue, postpone increase of other contacts, and add to the amount of the proxy overload. In the present work, while using UDP as transport layer protocol, invite retransmission timer (T1) was appropriately regulated and SIP functionality was improved. Therefore, by proposing an adaptive timer of invite message retransmission, attempts were made to improve the time of session initiation and consequently improve the performance. Performance of the proposed SIP was implemented and evaluated by SIPP software in a real network environment and its accuracy and performance were demonstrated

High Load Control Mechanism for SIP Servers

To start voice, image, instant messaging, and generally multimedia communication, session communication must begin between two participants. SIP (session initiation protocol) that is an application layer control induces management and terminates this kind of sessions. As far as the independence of SIP from transport layer protocols is concerned, SIP messages can be transferred on a variety of transport layer protocols including TCP or UDP. Mechanism of Retransmission that is embedded in SIP could compensate for the missing packet loss, in case of need. This mechanism is applied when SIP messages are transmitted on an unreliable transmission layer protocol like UDP. Also, while facing SIP proxy with overload, it could cause excessive filling of proxy queue, postpone increase of other contacts, and add to the amount of the proxy overload. In the present work, while using UDP as transport layer protocol, invite retransmission timer (T1) was appropriately regulated and SIP functionality was improved. Therefore, by proposing an adaptive timer of invite message retransmission, attempts were made to improve the time of session initiation and consequently improve the performance. Performance of the proposed SIP was implemented and evaluated by SIPP software in a real network environment and its accuracy and performance were demonstrated

High Load Diminution by Regulating Timers in SIP Servers

To start voice, image, instant messaging, and generally multimedia communication, session communication must begin between two participants. SIP (session initiation protocol) that is an application layer control induces management and terminates this kind of sessions. As far as the independence of SIP from transport layer protocols is concerned, SIP messages can be transferred on a variety of transport layer protocols including TCP or UDP. Mechanism of Retransmission that is embedded in SIP could compensate for the missing packet loss, in case of need. This mechanism is applied when SIP messages are transmitted on an unreliable transmission layer protocol like UDP. Also, while facing SIP proxy with overload, it could cause excessive filling of proxy queue, postpone increase of other contacts, and add to the amount of the proxy overload. In the present work, while using UDP as transport layer protocol, invite retransmission timer (T1) was appropriately regulated and SIP functionality was improved. Therefore, by proposing an adaptive timer of invite message retransmission, attempts were made to improve the time of session initiation and consequently improve the performance. Performance of the proposed SIP was implemented and evaluated by SIPP software in a real network environment and its accuracy and performance were demonstrated

Analysis of Session Establishment Signaling Delay in IP Multimedia Subsystem

Abstract. This paper investigates and analyzes SIP delay in the session establishment signaling procedure in the IMS system. We investigate the delay for end-to-end link scenarios such as WiMAX-to-WiMAX, UMTS-to-UMTS, UMTS-to-WiMAX and vice versa. The analyses consider three types of delays: transmission delay, processing delay and queuing delay. The obtained results show that the main delay of session establishment signaling process is due to the processing delay. In addition, the lower channel rate in the UMTS network as well as IMS service rate has significant impact to the session establishment signaling delay

A study of mobile VoIP performance in wireless broadband networks

Voice service is to date still the killer mobile service and the main source for operator revenue for years to come. Additionally, voice service will evolve from circuit switched technologies towards packet based Voice over IP (VoIP). However, using VoIP over wireless networks different from 3GPP cellular technologies makes it also a disruptive technology in the traditional telecommunication sector. The focus of this dissertation is on determining mobile VoIP performance in different wireless broadband systems with current state of the art networks, as well as the potential disruption to cellular operators when mobile VoIP is deployed over different access networks. The research method is based on an empirical model. The model and experiments are well documented and based on industry standards for voice quality evaluation. The evaluation provides results from both experiments in a controlled laboratory setup as well as from live scenarios. The research scope is first, evaluate each network technology independently; second, investigate vertical handover mobility cases; third, determine other aspects directly affecting end user experience (e.g., call setup delay and battery lifetime). The main contribution of this work is a systematic examination of mobile VoIP performance and end user experience. The research results point out the main challenges for achieving call toll quality, and how derive the required changes and technological performance roadmap for improved VoIP service. That is, investigate how the performance and usability of mobile VoIP can eventually be improved to be a suitable substitute for circuit switched voice. In addition, we evaluate the potential disruption to cellular operators that mobile VoIP brings when deployed over other access networks. This research extends the available knowledge from simulations and provides an insight into actual end user experience, as well as the challenges of using embedded clients in handheld devices. In addition, we find several issues that are not visible or accounted for in simulations in regard to network parameters, required retransmissions and decreased battery lifetime. The conclusion is that although the network performance of several wireless networks is good enough for near toll quality voice in static scenarios, there are still a number of problems which make it currently unfeasible to use as a primary voice service. Moreover, under mobility scenarios performance is degraded. Finally, there are other issues apart from network performance such as energy consumption, hardware limitations and lack of supporting business models (e.g., for WiFi mesh) that further limit the possibility of rolling out mobile VoIP services

Private Communication Detection via Side-Channel Attacks

Private communication detection (PCD) enables an ordinary network user to discover communication patterns (e.g., call time, length, frequency, and initiator) between two or more private parties. Analysis of communication patterns between private parties has historically been a powerful tool used by intelligence, military, law-enforcement and business organizations because it can reveal the strength of tie between these parties. Ordinary users are assumed to have neither eavesdropping capabilities (e.g., the network may employ strong anonymity measures) nor the legal authority (e.g. no ability to issue a warrant to network providers) to collect private-communication records. We show that PCD is possible by ordinary users merely by sending packets to various network end-nodes and analyzing the responses. Three approaches for PCD are proposed based on a new type of side channels caused by resource contention, and defenses are proposed. The Resource-Saturation PCD exploits the resource contention (e.g., a fixed-size buffer) by sending carefully designed packets and monitoring different responses. Its effectiveness has been demonstrated on three commercial closed-source VoIP phones. The Stochastic PCD shows that timing side channels in the form of probing responses, which are caused by distinct resource-contention responses when different applications run in end nodes, enable effective PCD despite network and proxy-generated noise (e.g., jitter, delays). It was applied to WiFi and Instant Messaging for resource contention in the radio channel and the keyboard, respectively. Similar analysis enables practical Sybil node detection. Finally, the Service-Priority PCD utilizes the fact that 3G/2G mobile communication systems give higher priority to voice service than data service. This allows detection of the busy status of smartphones, and then discovery of their call records by correlating the busy status. This approach was successfully applied to iPhone and Android phones in AT&T's network. An additional, unanticipated finding was that an Internet user could disable a 2G phone's voice service by probing it with short enough intervals (e.g., 1 second). PCD defenses can be traditional side-channel countermeasures or PCD-specific ones, e.g., monitoring and blocking suspicious periodic network traffic

Modélisation et caractérisation du trafic IMS

IMS : architecture et services -- Architecture de services -- Évaluation de performance